1. Technical Field
The present invention relates in general to a system and method for improved quality signal re-sampling. More particularly, the present invention relates to a system and method for adding a weighed signal average to a re-sampled signal in order to reduce the re-sampled signal's output signal error.
2. Description of the Related Art
The advancement of the electronics industry has brought about an advancement of digital audio. Digital audio is an audio signal that is represented in digital format whereby the digital format includes a corresponding sample rate. A digital audio signal's sample rate is the number of samples the signal provides per second in order to represent an audio signal. The more samples per second a signal provides, the more accurate the digital representation of the audio signal. For example, the current sample rate for CD-quality audio is 44,100 samples per second, which reproduces audio frequencies up to 20,500 hertz.
In addition to CD quality audio, other digital audio formats are becoming industry standards. For example, DAT (Digital Audio Tape) is a standard medium and technology for digitally recording audio on tape at a professional quality level. For example, professional and semi-professional recording studios use DAT to archive master recordings. A DAT drive is a digital tape recorder with rotating heads similar to those found in a video deck, and typically record at sample rates of 48,000 samples per second.
Since multiple digital audio standards exist, users may wish to convert, or re-sample a digital audio signal from one format to another format. For example, a user may wish to convert a CD digital audio signal at 44,100 samples per second to a DAT digital audio signal at 48,000 samples per second. When performing digital audio re-sampling, a decision is made as to how many input samples to use when generating an output sample. In theory, a user may use an infinite amount of input samples to reproduce an output sample, which results in no output signal error. In reality, a user determines a finite number of input samples to use when generating an output signal, which, in turn, induces signal error.
A challenge found in determining how many input samples to use, however, is that consequences result, regardless of a user's sample quantity choice. If a user chooses a small input sample quantity, the output signal accrues a large signal error from clipping the input samples to a small finite number. On the other hand, if a user chooses a large sample quantity, the output signal accrues a small signal error, but the re-sampling process take a tremendous amount of memory and processing power.
What is needed, therefore, is a system and method to use a small input sample quantity to re-produce a quality digital audio signal by minimizing the re-produced signal's error.